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Nureva Granted a Fourth Patent for its Microphone Mist Technology

Nureva has announced that the U.S. Patent and Trademark Office has issued patent number 10,499,151 for embedding additional information in a sound mask noise signal.
(Image credit: Nureva)

Nureva has announced that the U.S. Patent and Trademark Office has issued patent number 10,499,151 for embedding additional information in a sound mask noise signal. This is the company’s fourth patent for its Microphone Mist technology platform, which powers its HDL300 audio conferencing systems and the HDL200 system that was announced earlier this week. 

The patent describes the ability of the microphone and speaker system to create a continuous sound mask signal that enables an impulse response measurement for each microphone and speaker combination in real time. This allows the system to instantly auto-calibrate the multichannel echo canceler for each speaker to microphone combination and provide continuous, real-time, in-room impulse response measurements. 

Nureva audio systems maintain consistent performance by automatically and continuously adapting to changes in room configurations or a change in the location of the Nureva system itself. For example, no manual recalibration is required by IT staff or a service technician when furniture or mobile walls are moved to accommodate varying meeting scenarios. 

“Our family of Microphone Mist technology patents reflects Nureva’s continued investment in a rich and multilayered platform with plenty of runway for future developments,” said Nancy Knowlton, Nureva’s CEO. “The acknowledgement of our inventions by the USTPO is very exciting as we continue to explore unique ways of making the lives of our customers easier and more productive.”

This patent is the latest addition to Nureva’s active and growing intellectual property portfolio in support of its Microphone Mist technology platform, and it comes on the heels of three other patents announced in the fall last year. 

Microphone Mist technology uses sophisticated algorithms to simultaneously process sound from all virtual microphones to provide remote participants with a high-quality listening experience, enabled by continuous auto-calibration, simultaneous echo cancellation, position-based automatic gain control, and sound masking.